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...tion67 = gam1*SES Alienation71 = gam2*SES + beta*Alienation67 V(Anomia67) = the1 V(Anomia71) = the1 V(Powerless67) = the2 V(Powerless71) = the2 V(SES) = phi C(Anomia67, Anomia71) = the5 C(Powerless67, Powerless71) = the5 Similarly, the following are equivalent specifications of a CFA model for the Thurstore mental-tests data: (1) in CFA format: mod.cfa.thur.c <- cfa() FA: Sentences, Vocabulary, Sent.Completion FB: First.Letters, 4.Letter.Words, Suffixes FC: Letter.Series, Pedigrees, Letter.Group cfa.thur.c <- sem(mod.cfa.thur.c, R.thur, 213) summary(cfa.thur.c) (2) in equation format: m...

...tion67 = gam1*SES Alienation71 = gam2*SES + beta*Alienation67 V(Anomia67) = the1 V(Anomia71) = the1 V(Powerless67) = the2 V(Powerless71) = the2 V(SES) = phi C(Anomia67, Anomia71) = the5 C(Powerless67, Powerless71) = the5 Similarly, the following are equivalent specifications of a CFA model for the Thurstore mental-tests data: (1) in CFA format: mod.cfa.thur.c <- cfa() FA: Sentences, Vocabulary, Sent.Completion FB: First.Letters, 4.Letter.Words, Suffixes FC: Letter.Series, Pedigrees, Letter.Group cfa.thur.c <- sem(mod.cfa.thur.c, R.thur, 213) summary(cfa.thur.c) (2) in equation format: m...

...have ambiguity between extensions. Are you using the '.' wildcard? Richard -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New t= o Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www....

Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful.

Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Cheers, Richard. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060109/26c4a63c/attachment.htm

...have ambiguity between extensions. Are you using the '.' wildcard? Richard -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New t= o Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www....

...have ambiguity between extensions. Are you using the '.' wildcard? Richard -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New t= o Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www....

Hi All, I have installed asteriskNow with Asterisk 1.6.2.11 and FreePBX 2.7.0.10. I have configured x100p fxo card in my asterisk box. But in my cli mode i am not getting the command *"dahdi show status"* Output of CLI : astrisks*CLI> *dahdi show status* No such command 'dahdi show status' (type 'core show help dahdi show' for other possible commands) I

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -------------- next

Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register => 908366554:396444 at carrier.jazzey.com register => 908366554:396444 at sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444

List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer

...have ambiguity between extensions. Are you using the '.' wildcard? Richard -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New t= o Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www....

hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . ?service zaptel restart? or there is any other command Thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username

Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110621/764a6fa9/attachment.htm>

...y several people, an Overview document has been posted at http://blogs.sun.com/sunay The reference section also contains a pointer to the slides that will be used for tonights presentation. See you all there. Cheers, Sunay -------- Original Message -------- Subject: [osol-discuss] SVOSUG - This Thurs, August 24, Sunay Tripathi presents Crossbow 7:30pm SCA03 Date: Mon, 21 Aug 2006 11:35:26 -0700 From: Alan DuBoff <Alan.DuBoff at sun.com> Reply-To: Alan DuBoff <Alan.DuBoff at sun.com> Organization: Solaris x86 Engineering To: Alan DuBoff <Alan.DuBoff at sun.com> CC: opensolaris...

...; Sonny. >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>>> http://www.asterisk.org/hello >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>&gt...

Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111214/b462516a/attachment.htm>

Hi all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI |